similar to: [1.4/AGI] CHANNEL STATUS never "down & available"?

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

Hello Incoming calls from the FXO trigger an AGI script which simply NOOP data sent by Asterisk through stdin. The first two NOOP work fine, but after this, Asterisk isn't happy: ============= extensions.conf ... [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/test.lua) exten => s,n,Wait(5) exten => s,n,Hangup =============

Hello I'm taking a look at how to write scripts to be called from the dialplan, and saw pbx_lua mentioned. I'd like to know more about this feature, such as what the difference is with just calling the Lua interpreter through AGI (same difference as between php-cgi and mod_php?), whether it's production-ready, etc. Thank you for any help.

Hello The following, dead simple Bash script ran as AGI doesn't reply to Asterisk: ============= extensions.conf [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/basic.agi) exten => s,n,Hangup() ============= /var/tmp/basic.agi #!/bin/bash #Ripped from #http://lists.digium.com/pipermail/asterisk-users/2003-July/008554.html while

Hello, I'm trying to call different SIP-accounts to connect them to a conference. This is my call-file : Channel: SIP/test3&SIP/test1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: from-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3&SIP/test1 for 1000 at from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]:

Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I need Asterisk (1.4.39.2) to simulate a flash hook (ie. hitting the "R" key on European handsets) so I can put a call on hold, dial a second number, and set up a conference call. By default, linux/include/dahdi/kernel.h sets the flashtime to 750ms, which appears to be too long for European telcos, as they seem to expect a line cut of about 100ms. After editing the

Hello I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? Ideally, I'd rather use a rich language like PHP or Python, but can those be fit with even their common modules into such small

I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the "avoided deadlock" message below. *CLI> == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on 'SIP/3211-1-081c40a8' -- Executing NoOp("SIP/3211-1-081c40a8", "") in new stack -- Executing AGI("SIP/3211-1-081c40a8", "diallocal.agi") in new

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

Hi Just installed Asterisk on a Debian Woody/testing. I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago). The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory: the test.call file looks like this: #Simple test call script. #call my

I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a

not able to run my php from AGIi am using asterisk 1.8.13 (debian)i am able to make call file using php command line..but when executing php from AGI, it is not working..kindly see the attachment if bellow text is not readable...___________________________________________________ File: /etc/asterisk/extensions.conf[call]exten => call,1,Answerexten => call,n,Playback(hello-world)exten =>

I can make calls over a SIP trunk as SIP/<trunk>/number I am trying to make calls over an extension thought using the same format SIP/4452/number - its not working. person says they can connect a software as extension 4452 and it works just fine. I have my register: register => 4452 at X.X.X.X/4452 [4452] type=friend username=4452 host=X.X.X.X allow=all dtmfmode=inband When I try to

Hey guys I was hoping I could get a few pointers on a problem I have been trying to debug for the last couple of months regarding asterisk AGI scripts and unexpected termination. I have this agi script that accepts incoming faxes using RxFax on the latest asterisk 1.4 branch. Its written with perl and it works fine except for one line that causes the entire script to terminate unexpectedly. The

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently

Hello I'm pretty much a newbie when it comes to C, but I have to use this language to write a couple of AGI proggies because I need them to be statically compiled. Strangely enough, Google didn't return much when looking for the "Hello, world!" of AGI in C. The following doesn't work: The file never gets written: =========== //check_cid.c #include <stdio.h> #include

-- jand. more than just a group Asterisk AGI Command GET DATA is usually of this form GET DATA timeout max_digits When I execute this command, I get only a single digit, regardless of what the value of max_digits is, Also the script quits Immediately after the press of the digit regardless of what the value of timeout is, This is really un-desirable as I will like to GET multiple DTMF digits

Dear All, I'm trying to place some calls using php-agi, the problem is, i'm sending the call to one of my providers via sip, if the leg 2 hangup the asterisk don't end the call. I'm using the dial command as follow: SIP/destination at provider|60|HRrL(120000:61000:30000) Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there, Im getting this notice in CLI, but the call quality is okey, Im using digium TE406 and asterisk 1.2.4. here are the CLI actual logs: -- Executing SetAccount("Local/50015308467418@default-ca2e,2", "XXXXXX") in new stack -- Executing AGI("Local/50015308467418@default-ca2e,2", "call_log.agi|50015308467418") in new stack -- Launched AGI